Пожалуйста, войдите здесь. Часто задаваемые вопросы О нас
Задайте Ваш вопрос

не работают звонки с asterisk на cisco 5350

0

С АТС-cisco-* звонок проходит , а вот в обратном направлении к сожалению нет. Товарищи , уважаемые , помогите разобрать дебаг с cisco:

**GW#debug ccsip calls**
SIP Call statistics tracing is enabled
GW#
Jul 12 13:16:49: //297806/86E4F67583B7/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x6588B2E4
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 4010111
Called Number            : astral
Source IP Address (Sig  ): 10.241.2.240
Destn SIP Req Addr:Port  : 10.241.1.215:5060
Destn SIP Resp Addr:Port : 10.241.1.215:5060
Destination Name         : 10.241.1.215

GW#
Jul 12 13:16:49: //297806/86E4F67583B7/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101
Source IP Address (Media): 10.241.2.240
Source IP Port    (Media): 16866
Destn  IP Address (Media): 10.241.1.215
Destn  IP Port    (Media): 19060
Orig Destn IP Address:Port (Media): 0.0.0.0:0

Jul 12 13:16:49: //297806/86E4F67583B7/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 500



____________________________________________________________________

**GW#debug ccsip all**

INVITE sip:astral@10.241.2.240 SIP/2.0
Via: SIP/2.0/UDP 10.241.1.215:5060;branch=z9hG4bK1a90413a;rport
Max-Forwards: 70
From: "4010111" <sip:4010111@10.241.1.215>;tag=as072bf4c5
To: <sip:astral@10.241.2.240>
Contact: <sip:4010111@10.241.1.215:5060>
Call-ID: 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.0
Date: Thu, 12 Jul 2012 09:06:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 729095750 729095750 IN IP4 10.241.1.215
s=Asterisk PBX 1.8.13.0
c=IN IP4 10.241.1.215
t=0 0
m=audio 11818 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/State/sipSPIChangeState: 0x634D9914 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.241.1.215,Port 5060, Transport 1, SentBy Port 5060
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone UA to SIP default timezone = GMT
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.241.1.215,Port 5060, Transport 1, SentBy Port 5060
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetGtdBody: No valid GTD body found.
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x634D9914 key=08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060astral
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: astral
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: 4010111
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name 4010111, number 4010111, Calling oct3 0x00, oct_3a 0x80, Called number astral
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIGetCallConfig: Peer tag 4010 matched for incoming call
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIGetCallConfig: Not using Voice Class Codec
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPICopyPeerDataToCCB:
From CLI: Modem NSE payload = 100, Passthrough = 0, Modem relay = 0, Gw-Xid = 1
SPRT latency 200, SPRT Retries = 12, Dict Size = 1024
 String Len = 32, Compress dir = 3
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIContinueNewMsgInvite: Calling name 4010111, number 4010111, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number astral, oct3 0x00
Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:20
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :20, codecbytes: 160
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved.
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoDTMFRelayNegotiation: Payload type (101) is reserved for requested dtmf relay mode.
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named event(NE) match in fmtp list of events.
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sip_do_nse_negotiation: SDP not present. Use local NSE payload 100.
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
        payload_type=8, codec_bytes=160, codec=g711alaw, dtmf_relay=rtp-nte
        stream_type=voice+dtmf (1), dest_ip_address=10.241.1.215, dest_port=11818
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Media/sipSPIUpdCallWithSdpInfo:
        Preferred Codec        : g711alaw, bytes :160
        Preferred  DTMF relay  : sip-notify
        Preferred NTE payload  : 101
        Early Media            : No
        Delayed Media          : No
        Bridge Done            : No
        New Media              : No
        DSP DNLD Reqd          : No

Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.241.2.240
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_report_media_to_peer:
 callId 287855 peer 0 flags 0x201
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CallID 287855, sdp 0x6724C3A4 channels 0x634DB1D4
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 8 mline 1
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecing codec g711alaw
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :20, codecbytes: 160
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 6 ptype 8 time 20, bytes 160  as channel 0 mline 1 ss 0 10.241.1.215:11818
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 101 mline 1
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_report_media_to_peer:
Report initial call media
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/copy_channels:
 callId 287855 size 80 ptr 0x631EC078)
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_report_media_to_peer:
CCSIP: Unable to report channel ind
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Media/sipSPIUpdCallWithSdpInfo:
          Stream type            : voice+dtmf
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Callid                 : -1
          Negotiated Codec       : g711alaw, bytes :160
          Negotiated DTMF relay  : rtp-nte
          Negotiated NTE payload : 101
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : 10.241.2.240:0
          Media Dest Addr/Port   : 10.241.1.215:11818

Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIHandleInviteMedia:
Negotiated Codec        : g711alaw, bytes :160
Preferred Codec         : g711alaw, bytes :160
Preferred  DTMF relay 1 : 8
Preferred  DTMF relay 2 : 6
Negotiated DTMF relay   : 6
Preferred and Negotiated NTE payloads: 101 101
Preferred and Negotiated NSE payloads: 100 100
Preferred and Negotiated Modem Relay: 0 0
Preferred and Negotiated Modem Relay GwXid: 1 0

Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoQoSNegotiation: SDP body with media description
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17558 for stream 1
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=17558
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 17558
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Negotiated method of dtmf relayand pyld: 6 101

Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed.
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x634D9914 key=08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060F7BF92C-5AA
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 4646F to table
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipSPITransportSendMessage: msg=0x65447538, addr=10.241.1.215, port=5060, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x00000000
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x65447538 to default port=5060
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: Connection obtained...sending msg=0x65447538
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x65447538, addr=10.241.1.215, port=5060, connId=1 for UDP
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/State/sipSPIChangeState: 0x634D9914 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_RECD_INVITE, SUBSTATE_NONE)
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIProcessContactInfo: Previous Hop 10.241.1.215:5060
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/act_recdinvite_disconnect: Performing disconnect
Jul 12 12
GW_ADPSU#:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipSPITransportSendMessage: msg=0x65445C78, addr=10.241.1.215, port=5060, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x6151A678
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x65445C78 to default port=5060
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: Connection obtained...sending msg=0x65445C78
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x65445C78, addr=10.241.1.215, port=5060, connId=1 for UDP
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sentErrResDisconnecting: Sent an 3456XX Error Response
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/State/sipSPIChangeState: 0x634D9914 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.241.1.215:5060;branch=z9hG4bK1a90413a;rport
From: "4010111" <sip:4010111@10.241.1.215>;tag=as072bf4c5
To: <sip:astral@10.241.2.240>;tag=F7BF92C-5AA
Date: Thu, 12 Jul 2012 09:06:19 GMT
Call-ID: 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0



Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.241.1.215:5060;branch=z9hG4bK1a90413a;rport
From: "4010111" <sip:4010111@10.241.1.215>;tag=as072bf4c5
To: <sip:astral@10.241.2.240>;tag=F7BF92C-5AA
Date: Thu, 12 Jul 2012 09:06:19 GMT
Call-ID: 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=16
Content-Length: 0



Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.241.1.215:5060
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x655522E0
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIFindCcbUASReqTable: *****CCB found in UAS Request table. ccb=0x634D9914
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:astral@10.241.2.240 SIP/2.0
Via: SIP/2.0/UDP 10.241.1.215:5060;branch=z9hG4bK1a90413a;rport
Max-Forwards: 70
From: "4010111" <sip:4010111@10.241.1.215>;tag=as072bf4c5
To: <sip:astral@10.241.2.240>;tag=F7BF92C-5AA
Contact: <sip:4010111@10.241.1.215:5060>
Call-ID: 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.0
Content-Length: 0



Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.241.1.215,Port 5060, Transport 1, SentBy Port 5060
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone UA to SIP default timezone = GMT
Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.241.1.215,Port 5060, Transport 1, SentBy Port 5060
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIIcpifUpdate: CallState: 2 Playout: 0 DiscTime:25978296 ConnTime 0
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/State/sipSPIChangeState: 0x634D9914 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to (STATE_DEAD, SUBSTATE_NONE)
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x634D9914
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 4010111
Called Number            : astral
Source IP Address (Sig  ): 10.241.2.240
Destn SIP Req Addr:Port  : 10.241.1.215:5060
Destn SIP Resp Addr:Port : 10.241.1.215:5060
Destination Name         : 10.241.1.215

Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101
Source IP Address (Media): 10.241.2.240
Source IP Port    (Media): 17558
Destn  IP Address (Media): 10.241.1.215
Destn  IP Port    (Media): 11818
Orig Destn IP Address:Port (Media): 0.0.0.0:0

Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 500

Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteCcbFromUASReqTable: ****Deleting from UAS Request table.
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x634D9914 key=08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060astral
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteCcbFromUASRespTable: ****Deleting from UAS Response table.
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x634D9914 key=08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060F7BF92C-5AA
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 4646F
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 634D9914
удалить закрыть спам изменить тег редактировать

спросил 2012-07-12 14:22:19 +0400

evers Gravatar evers
1 2 2

обновил 2012-07-12 15:16:54 +0400

zzuz Gravatar zzuz flag of Russian Federation
6744 2 6 69
http://line24.ru/

Comments

вербос с астериска можете показать?

telefonist ( 2012-07-13 14:48:55 +0400 )редактировать

достаточно

SIP/2.0 500 Internal Server Error

там видимо просто кривая настройка. Автор просто забил на чтение мануала, как это чаще всего бывает.

zzuz ( 2012-07-13 14:52:26 +0400 )редактировать

Очень на это похоже!

telefonist ( 2012-07-13 15:32:51 +0400 )редактировать

мануала ? а есть нормальный по сопряжению с cisco ? вот привожу конфиг
on ASTERISK
===sip.conf===
[astral]
type = friend
host = 10.241.2.240
username = astral
secret = astral!
qualify = yes
context = internal
disallow = all
allow = ulaw
insecure = port,invite
canreinvite = no
dtmfmode = rfc2833
language = ru

evers ( 2012-09-24 02:31:58 +0400 )редактировать

====extensions.conf===

[globals]
cisco = SIP/astral

[internal]
exten => _63XX,1,Dial(${cisco}) внутренние номера атс и циско
exten => _63XX,n,Hangup()

=====на CISCO====
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server
no call service stop

sip-ua
authentication username astral password astral!
registrar ipv4:10.241.1.215:5060 expires 3600
sip-server ipv4:10.241.1.215

evers ( 2012-09-24 02:33:51 +0400 )редактировать

ну и диалпир с циски на астериск

dial-peer voice 4010 voip
tone ringback alert-no-PI
destination-pattern 4010...
progress_ind setup enable 3
voice-class sip rel1xx disable
session protocol sipv2
session target ipv4:10.241.1.215:5060
session transport udp
dtmf-relay sip-notify rtp-nte h245-signal h245-alphanumeric
codec g711ulaw
no vad

evers ( 2012-09-24 02:34:19 +0400 )редактировать

ткните в мануал пожалуйста если он нормальный существует , а то гугл много знает , чересчур много )))

evers ( 2012-09-24 02:36:12 +0400 )редактировать

маршрутизация на AS5350 у вас настроена? можете выложить описание pots пиров?

brost ( 2013-03-15 15:20:36 +0400 )редактировать

всего то 6 месяцев прошло.

zzuz ( 2013-03-15 15:29:28 +0400 )редактировать

да. я уже понял. когда написал. вредно на ночь читать форум ))

brost ( 2013-03-18 10:00:00 +0400 )редактировать

Будьте первым, кто ответит на этот вопрос!

Please start posting your answer anonymously - your answer will be saved within the current session and published after you log in or create a new account. Please try to give a substantial answer, for discussions, please use comments and please do remember to vote (after you log in)!
[скрыть предварительный просмотр]

Закладки и информация

Добавить закладку

подписаться на rss ленту новостей

Статистика

Задан: 2012-07-12 14:22:19 +0400

Просмотрен: 950 раз

Обновлен: Jul 12 '12

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.